SIPjs + Asterisk > on Debian (updated)

This article willshow how to setup, install and deploy asterisk in Debian, and use the SIPjs by implementing the owo-phone example, unfortunatelly only works for following Debian versions: wheeze, jessie, strecht, for squeeze and lenny does not work due lack of resources (a hard disk and a powered machine) to make available.

IMPORTANT this will install asterisk from debian
backport if available, in wheeze will install 11.14 that has limited
support for streaming, in jessie will install asterisk 12 that have good
support, but if vegnuli repo are enable, wil install in jessie asterisk
13.14 with complete streaming support. For most modern Debian will
install lasted asterisk that have good complete streaming support.

2. Configuration asterisk

Generates a selft signed certificate for the wss entry point, configure http and sip modules for webrtc >

3. Configure SIPjs

Here we have a problem, chrome/like browsers dont allow easyle to setup a exception to your new ws entry
point self signed certificate, so maybe its recommended to use firefox/palemoon here next, for that, navigate
to https://127.0.0.1:8089/ws and add the certificate exception by click on the @avanced@ button at the screen advertise.Install apache, git, curl nodejs and npm

Open your browser and try to start communication
Now point your browser to http://localhost/sipjs/
Press the "wheel" button next to the "call" button and provide all the credentials
as well as the "ws" service entry point uri to be able to use it.

push the whell button aside to the "Call" button

use the sip.conf configure device name 1001 or 1060 can be

uri are the 1060@127.0.0.1 or 1001@127.0.0.1 or @ like 1060@10.101.10.222

auth name : 1060

password of 1060 "12345" as was paste in the sip.conf

ws server: wss://127.0.0.1:8089/ws or use ipaddress like wss://10.101.10.222:8089/ws

Once this way you can make a "call" to a number of the same "ws" WebRTC service
on that server prevously placed/configured, that means, you can perform voice
streaming chat between two telephones, but at least one will be digital, the browser phone,
by example the "1060" device/number can call the "1001" if each one configured in
different browsers, or any of those can make a call to the "1111" demo configured in extensions.